[cisco-voip] Please Help - Inbound dialing rings busy /Outbound works

ATIENZA, Gonzalo Gonzalo.ATIENZA at LA.LOGICALIS.COM
Tue Mar 22 11:41:13 EDT 2011


Good day Jay,

 

For what I´m seeing on the debugs on the inbound call the invite is sending an empty called number:

 

Received:

INVITE sip:5856786019 at 66.66.214.216:60954 SIP/2.0

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport

From: "+15856944923" ;tag=as2ccf6d86

To:

Contact:

 

That´s why you are sending:

 

Sent:

SIP/2.0 400 Bad Request - 'Invalid Host'

Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport

From: "+15856944923" ;tag=as2ccf6d86

To: ;tag=1A0D2FC-D89

 

Is that coming from the provider?  Check that out with them...

 

I don't have good experiences with NAT and anything related to voice, but with the NAT configuration you´ve got, packets are at least getting to the CME...  You could have one way audio issues after solving the empty called number problem.  In that case you might need to configure inspection to solve that problem, but I´m not sure if that would work with PAT.  Check this doc (it´s for an ASA):  http://www.cisco.com/en/US/docs/security/asa/asa82/configuration/guide/inspect_voicevideo.html#wp1204403

 

Regards.

                                   

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jay Stants
Sent: martes, 22 de marzo de 2011 11:53 a.m.
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Please Help - Inbound dialing rings busy /Outbound works

 

Good day everyone,

 

Still trying to figure this issue out. here is a summary of posts .. Some say that it's possibly an issue of Nat and Sip not liking each other, so i will try to be as detailed as possible in hopes that someone can see something i'm possibly missing. 

 

Attached is the lab design for the environment - I have both EIGRP and OSPF running and i redistribute both at the core switch - all routes are reachable

 

WAN router is a 2611XM with the following ACL's Defined

 

ip nat inside source list NAT_ACL interface FastEthernet0/1 overload 

! 

ip access-list standard NAT_ACL 

 permit 10.0.0.0 0.255.255.255 

! 

ip access-list extended FW_inbound 

 permit tcp any any established 

 permit udp any any 

 permit icmp any any echo-reply 

 deny   ip any any 

! 

 

CCME is configured on a 2811 router with a 16 port PoE module (NME-16ES-1G-P) - recently upgraded to IOS version AdvEnterpriseK9-M 15.1(2)GC / CCME 8.1

Configs are attached for review.

 

snet-voip1#sh sip-ua register status
Line                                                                     peer                 expires(sec)         registered         P-Associ-URI
================================ ========== ============ ========== ============
122812_cme                                                         -1                         82                     yes
5856786019                                                         20004                   101                    no

 

Debugs that i have run indicating that inbound there is an issue either from voip.ms to me or else on the CCME device with routing the incoming call to a phone. I'm not 100% positive where the issue lies. but dialing my DID just rings busy almost immediatly.(not a fast busy)

 

debug voip ccapi inout and debug ccsip messages

Mar 21 03:18:49.669: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:newyork.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK340EF7
From: ;tag=1A0CE20-114C
To:
Date: Mon, 21 Mar 2011 03:18:49 GMT
Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1300677529
CSeq: 380 REGISTER
Contact:
Expires: 180
Supported: path
Content-Length: 0

Mar 21 03:18:49.721: //417/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK340EF7;received=10.1.10.2
From: ;tag=1A0CE20-114C
To:
Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900
CSeq: 380 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Su
snet-voip1#pported: replaces
Contact:
Content-Length: 0

Mar 21 03:18:49.725: //417/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK340EF7;received=10.1.10.2
From: ;tag=1A0CE20-114C
To: ;tag=as78e4178e
Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900
CSeq: 380 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="08644cbc"
Content-Length: 0

Mar 21 03:18:49.729: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:newyork.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3411244
From: ;tag=1A0CE20-114C
To:
Date: Mon, 21 Mar 2011 03:18:49 GMT
Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1300677529
CSeq: 381 REGISTER
Contact:
Expires: 180
Authorization: Digest username="122812_cme",realm="newyork.voip.ms",uri="sip:newyork.voip.ms:5060",response="81e0f1e1d74178c268f1aec698535939",nonce="08644cbc",algorithm=MD5
Content-Length: 0

Mar 21 03:18:49.781: //417/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3411244;received=10.1.10.2
From: ;tag=1A0CE20-114C
To:
Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900
CSeq: 381 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0

Mar 21 03:18:49.797: //417/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3411244;received=10.1.10.2
From: ;tag=1A0CE20-114C
To: ;tag=as78e4178e
Call-ID: D60F7CDF-526111E0-8002D2DF-336AE900
CSeq: 381 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 180
Contact: ;expires=180
Date: Mon, 21 Mar 2011 03:18:49 GMT
Content-Length: 0

Mar 21 03:18:50.905: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5856786019 at 66.66.214.216:60954 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport
From: "+15856944923" ;tag=as2ccf6d86
To:
Contact:
Call-ID: 73f32c6f3b912602300b0c1814fc7b28 at 74.63.41.218
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856944923" ;privacy=off;screen=no
Date: Mon, 21 Mar 2011 03:18:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 2881 2881 IN IP4 74.63.41.218
s=session
c=IN IP4 74.63.41.218
t=0 0
m=audio 19002 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Mar 21 03:18:50.909: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport
From: "+15856944923" ;tag=as2ccf6d86
To: ;tag=1A0D2FC-D89
Date: Mon, 21 Mar 2011 03:18:50 GMT
Call-ID: 73f32c6f3b912602300b0c1814fc7b28 at 74.63.41.218
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Mar 21 03:18:50.961: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5856786019 at 66.66.214.216:60954 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK4ab70570;rport
From: "+15856944923" ;tag=as2ccf6d86
To: ;tag=1A0D2FC-D89
Contact:
Call-ID: 73f32c6f3b912602300b0c1814fc7b28 at 74.63.41.218
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856944923" ;privacy=off;screen=no
Content-Length: 0

Mar 21 03:19:07.973: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:newyork.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK34220CC
From: ;tag=1A115A4-EB0
To:
Date: Mon, 21 Mar 2011 03:19:07 GMT
Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1300677547
CSeq: 456 REGISTER
Contact:
Expires: 180
Supported: path
Content-Length: 0

Mar 21 03:19:08.029: //418/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK34220CC;received=10.1.10.2
From: ;tag=1A115A4-EB0
To: ;tag=as47fbc031
Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900
CSeq: 456 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, S
snet-voip1#UBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="74aab3a6"
Content-Length: 0

Mar 21 03:19:08.033: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:newyork.voip.ms:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3438DE
From: ;tag=1A115A4-EB0
To:
Date: Mon, 21 Mar 2011 03:19:08 GMT
Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1300677548
CSeq: 457 REGISTER
Contact:
Expires: 180
Authorization: Digest username="122812_cme",realm="newyork.voip.ms",uri="sip:newyork.voip.ms:5060",response="6ecb543fe1f1bcb70940d5c129e062d4",nonce="74aab3a6",algorithm=MD5
Content-Length: 0

Mar 21 03:19:08.085: //418/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3438DE;received=10.1.10.2
From: ;tag=1A115A4-EB0
To: ;tag=as47fbc031
Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900
CSeq: 457 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="6de8b321"
Content-Length: 0

Mar 21 03:19:09.029: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK34220CC;received=10.1.10.2
From: ;tag=1A115A4-EB0
To: ;tag=as47fbc031
Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900
CSeq: 456 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="74aab3a6"
Content-Length: 0

Mar 21 03:19:09.085: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.10.2:5060;branch=z9hG4bK3438DE;received=10.1.10.2
From: ;tag=1A115A4-EB0
To: ;tag=as47fbc031
Call-ID: D62E9D9A-526111E0-8003D2DF-336AE900
CSeq: 457 REGISTER
User-Agent: VoIPMS/SERAST
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="6de8b321"
Content-Length: 0

 

debug voip ccapi error and debug ccsip error

Mar 21 02:00:37.434: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
Mar 21 02:00:37.434: //-1/D988D9A98006/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE
Mar 21 02:00:37.438: //-1/xxxxxxxxxxxx/CCAPI/cc_set_post_tagdata:
CALL_ERROR; Avlist Set Is Failed
snet-voip1#
Mar 21 02:01:00.734: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: via branch list is:
Mar 21 02:01:00.734: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: end of list
Mar 21 02:01:00.790: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: via branch list is:
Mar 21 02:01:00.790: //-1/xxxxxxxxxxxx/SIP/Error/debugPrintBranchList: end of list
Mar 21 02:01:00.790: //346/000000000000/SIP/Error/ccsip_api_register_result_ind: Message Code Class 4xx Method Code 100 received for REGISTER
Mar 21 02:01:01.734: //-1/xxxxxxxxxxxx/SIP/Error/sipSPILocateInviteDialogCCB: Could not find ccb for response
snet-voip1#
Mar 21 02:01:01.790: //-1/xxxxxxxxxxxx/SIP/Error/sipSPILocateInviteDialogCCB: Could not find ccb for response

 

If more detail is needed please let me know and i can provide additional debugs. I've done an Echo test on the sip provider side to the DID and that was successful and the provider sees the call delivered so i believe that to validate the config within the account on voip.ms side to be correct as far as recieving calls from PSTN and delivering to DID / SIP side. 

 

Any help is much appreciated

Regards,

 

Jay Stants

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20110322/a74e267b/attachment.html>


More information about the cisco-voip mailing list